Yes they did.
On 3/26/12, SamyGo wrote:
> Good to know, hope our replies did some help :)
>
> On Thu, Mar 22, 2012 at 7:39 PM, James Mutuku wrote:
>
>> Hi,
>>
>> Thanks for the support. Issue solved. Somehow the routes on the fxo
>> gw were not working.
>>
>>
>>
>> On 3/21/12, James Mutuku wro
Good to know, hope our replies did some help :)
On Thu, Mar 22, 2012 at 7:39 PM, James Mutuku wrote:
> Hi,
>
> Thanks for the support. Issue solved. Somehow the routes on the fxo
> gw were not working.
>
>
>
> On 3/21/12, James Mutuku wrote:
> > Hi,
> >
> > I have configured a route on the fxo
Hi,
Thanks for the support. Issue solved. Somehow the routes on the fxo
gw were not working.
On 3/21/12, James Mutuku wrote:
> Hi,
>
> I have configured a route on the fxo to send all incoming sip traffic
> to the "fxo" ports.
>
> I will try set the specific digits and see.
>
> On 3/21/12, Sa
Hi,
I have configured a route on the fxo to send all incoming sip traffic
to the "fxo" ports.
I will try set the specific digits and see.
On 3/21/12, SamyGo wrote:
> 404 NOT FOUND means that they were unable to find any
> destination/route/rule/prefix match corresponding to your dialled number.
404 NOT FOUND means that they were unable to find any
destination/route/rule/prefix match corresponding to your dialled number.
See your FXO gateway configuration Web-UI for outbound patterns OR verify
that the FXO has its outbound line configured and working properly.
On Wed, Mar 21, 2012 at 5:20
I am setting up asterisk->fxo gw.
404 Not Found (User not found) means the user is not found, but I
don't need to have extensions or authentication on the fxo gw
On 3/21/12, Michael L. Young wrote:
>> [0K
>> <--- SIP read from UDP:192.168.9.251:5060 --->
>> SIP/2.0 404 Not Found
>>
>> Via: SIP/2
> [0K
> <--- SIP read from UDP:192.168.9.251:5060 --->
> SIP/2.0 404 Not Found
>
> Via: SIP/2.0/UDP 192.168.9.250:5060;rport=5060;branch=z9hG4bK111ef687
>
> To:
> ;tag=1332328154302b4aa3-f15d-4eb7-beee-97782e7cbd06
>
> From: "pbxserver" ;tag=as66c75bd7
>
> CSeq: 102 INVITE
>
> Call-ID: 4ce934e
my sip traces are below
Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 192.168.9.250 port 17722
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.
I am still getting the same error
On 3/21/12, white hat wrote:
> Just a guess here, but it looks like you are dialing a 10 digit phone
> number but the dial pattern in your outbound route does not handle that.
>
> Try using a different dial pattern in your outbound route such as:
>
> 1NXXNXX
Just a guess here, but it looks like you are dialing a 10 digit phone
number but the dial pattern in your outbound route does not handle that.
Try using a different dial pattern in your outbound route such as:
1NXXNXX or 9|1NXXNXX
I believe that asterisk is telling you that all circuits
This is not in human readable format, but using my special powers I was
able to locate the lines
-- Called fxosip/0799490994
-- SIP/fxosip-0015 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
Please enable sip debug for this carrier and then try to send the sip
traces in
I have setup a trunk on freepbx and the outbound route. Everytime I
dial via the trunk, I get "all circuits are busy now". Incoming calls
are working fine on the trunk.
This is my dial
9|XXX.
and these are my peer details
allow=ulaw&alaw
canredirect=no
disallow=all
dtmfmode=rfc2833
host=19
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